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标准的10个数字和26个字母的男语音数据库，用于语音信号处理(10 numbers and 26 letters of the standard male voice database for speech signal processing)

这是基于HMM的数字语音识别，实现了0到9 的识别，并带有GUI界面，可以进行录音和播放，预处理，识别结果准确，并附有音频(This is a digital speech recognition based on HMM identification of 0-9, with a GUI interface and recording box player, there are pretreatment, accurate identification, together with audio)

HMM隐马尔科夫模型源码，实现模型的训练(Hidden Markov Model HMM source, the realization of model training)

辅助计算SDR 评测语音质量参数 评估语音质量的好坏(caculate SDR)

端点检测程序 多频带，熵 谱熵 teger能量 希望大家好好学习好好利用(endpoint detection procedures multiband, entropy spectral entropy teger energy hope that we learn to make good use)

语音信号的盲源分离算法研究及应用，本程序利用matlab实现(Blind Source Separation Algorithm Research and Application of the voice signal, the program using matlab realize)

线性预测倒谱系数（Linear Prediction Cepstrum Coefficient,LPCC）是线性预测系数（Linear Prediction Coefficient,LPC）在倒谱域中的表示。该特征是基于语音信号为自回归信号的值设，利用线性预测分析获得倒谱系数。(Linear Prediction Cepstrum Coefficient)

ADPCM(Adaptive Pluse Coding Modular)便解码器，供大家学习参考(ADPCM (Adaptive Pluse Coding Modular) decoder will be available for everyone to learn the reference)

基于LMS(最小均方误差算法)的自适应滤波 基于LMS(最小均方误差算法)的自适应滤波(based on the LMS (MMSE) algorithm based on the LMS adaptive filtering (minimum mean square error algorithm) the adaptive filtering based on the LMS (minimum mean square error algorithm) Adaptive Filter)

观测实时模拟信号（语音）的频谱 用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图。 (Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw the spectrum map and voice waveform.)